similar to: Asterisk doesn't relay remote MOH during hold

Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: ---------- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the

hi, its possibe to "dont start" music on hold when caller (from sip operator trunk) press HOLD (i.e. on mobile phone) Asterisk acts on SDP a=sendonly i want pass trough media from SIP trunk provider Marek

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am having a strange problem with MOH. Say I have two users, A and B. I can set MOH in the extension for B and if A calls B and B hits hold, A will hear B's hold music. If however A hits hold, it goes to the default music. If I pull the setmusiconhold from extensions.conf and use musicclass in sip.conf under the peer A, I get the same thing. Peer A has musicclass set and A calls B and B

Hi All, When asterisk bridges a call between 2 peers and peer-A's user puts the call on hold, then peer-A sends a INVITE with recvonly in the SDP. Asterisk responds to peer-A with sendonly in the SDP and asterisk sends an INVITE to peer-B with recvonly in the SDP. Peer-B then responds with a sendonly in the SDP. I've noticed in the above scenario that peer-B contiutes to send audio to

Hi, I've been trying to get MoH files to sound decent. I've got a hold of Royalty-free Classical music (a safe choice for most of my customers) and I`ve been trying to convert them to the normal telephony/Asterisk format using sox. Unfortunately, it sounds really bad. I don't expect concert hall quality of course, 8000KHz being what it is, but is there a better way to convert

Hi All, I have a problem with asterisk and call hold. In the re-invite package when I take the call to the hold, the SDP value "a=sendrecv" is present, according to the rfc3264 the sdp value a must be mark with "sendonly". I've already tried with Asterisk 1.8 and Asterisk 11, but there is the same problem. I've already read all the information about canreinvite and

Hi folks I'm not sure what I am missing but I cannot find a predefined channel variable to identify the SIP peer/user which has initiated a call and established the channel. The one option is to extract it from the CHANNEL variable, but that is fraught with difficulties. Is there another variable I don't know about or another way to do this? Thanks in advance! Richard -- Richard

asterisk v1.0.3, mpg123 v59r, shoutcast server. When first starting asterisk all is fine, moh/mpg processes start, can see asterisk client connections on shoutcast monitor as well and I've got mp3 streamed music on hold, cool! After aprx. 32-105 seconds the asterisk client connections close on the shoutcast server. The moh/mpg processes are still running, but are now just looping a

I'm working through my list of features people will expect, and Hold Pickup is at the top at the moment -- has anyone done any work on this? We've had some unpleasant experiences with call parking, and everyone seems to like the Hold Pickup model. If you don't know what I mean by Hold Pickup, it's sort of a reverse transfer; pick up the nearest phone and dial

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

Hi folks I have a handset talking to Asterisk, which in turn puts the call through to an ITSP. The handsets likes to send audio in 40ms frames (even though Asterisk requests 20ms frames with a ptime header in the SDP). The ITSP doesn't request any particular frame length (with ptime) or state a maximum length (with maxptime), so when Asterisk receives the 40ms media frames from the handset,

Hi list, I have a strange comportment of the MOH system on my asterisk. When i respond to a call and after fews second i set this call in hold mode the correspondent listen the music fine. When i re-take my correspondent at T0 instant the music is paused. And when i re-hold him at T60 (60 second later) the sound is always at T0 when he was stopped at T0. So the music is stopped and don't

Hi all, I've asterisk 1.2.5 , and what is happening is this: Sip user agent "A" calls a pstn "phone B" Sip User agent Activates MOH. "B" starts listening. "A" doesn't hangup and just Disconnect Sipoftphone XLITE (exit) "B" stills listenning Music on Hold and "A" has left Asterisk, who hangs the call? only when B hangs...

Hi, when I put a call on hold from my Nokia E51 (SIP client), the other side does NOT hear music on hold although sip debug / wireshark shows that the E51 tells the asterisk that it now holds the call. Canreinvite is set to "no". Also, when parking a call (features.conf), the parked caller does not hear music on hold. In queues, when using "#" and when using the hold

So, Tilghman has put a particularly useful patch in the bugtracker: streaming music-on-hold is now supported. You can now specify .mp3 streams to be played back as MOH in the various places where MOH is used. Hopefully, Mark will install into the main CVS tree shortly. http://bugs.digium.com/bug_view_page.php?bug_id=0000413 This allows you to use the very sophisticated mp3 streaming audio

Hello, I am having problems with music on hold on grandstream phones. When I press Hold button on grandstream phone this is the debug of sip. But nothing happens, no music. Is it problem of asterisk or grandstream budget phone? Sip read: INVITE sip:1105@192.168.1.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK7fcd3a44e7721b41 From:

finished hooking up the altivec stuff so it works in ProjectBuilder. I ran a test, doing a 'flac -t' on 400MB of files compresses at level 5. the runtime dropped from from 180 sec to 105 sec! once I get the latest autotools on my ibook I'll try and get asm compilation to work that way. Josh --- Josh Coalson <xflac@yahoo.com> wrote: > OK, checked it all in (only minor

Hello: I have read that music on hold requires a timing source (which I never had to worry about previously since the server had zaptel hardware in it)...now I'm configuring a server in a colo which has no zaptel hardware. If I use the dialplan to run MusicOnHold(), I do get the music upon dialling that extension, but if I try to use the hold button on either a 7960 or X-Lite I get

Howdy. I'm working on Altivec versions of some of the libFLAC functions. I figured the best candidates would be those that had MMX/SSE/3dnow versions, and I picked FLAC__lpc_restore_signal() to do first, since it's relatively simple. In stepping through some runs, it appears that 'order' mod 4 is always 0. Is that guaranteed, either by the format or by higher functions in the